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-rw-r--r--sound/soc/omap/omap-mcbsp.c175
-rw-r--r--sound/soc/omap/omap3pandora.c36
-rw-r--r--sound/soc/omap/rx51.c73
3 files changed, 205 insertions, 79 deletions
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 6f44cb4d30b8..86f213905e2c 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -59,6 +59,7 @@ struct omap_mcbsp_data {
int configured;
unsigned int in_freq;
int clk_div;
+ int wlen;
};
#define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id)
@@ -154,20 +155,51 @@ static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+ struct omap_pcm_dma_data *dma_data;
int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
- int samples;
+ int words;
+
+ dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
- samples = snd_pcm_lib_period_bytes(substream) >> 1;
+ /*
+ * Configure McBSP threshold based on either:
+ * packet_size, when the sDMA is in packet mode, or
+ * based on the period size.
+ */
+ if (dma_data->packet_size)
+ words = dma_data->packet_size;
+ else
+ words = snd_pcm_lib_period_bytes(substream) /
+ (mcbsp_data->wlen / 8);
else
- samples = 1;
+ words = 1;
/* Configure McBSP internal buffer usage */
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
+ omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, words);
else
- omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
+ omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, words);
+}
+
+static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *buffer_size = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct omap_mcbsp_data *mcbsp_data = rule->private;
+ struct snd_interval frames;
+ int size;
+
+ snd_interval_any(&frames);
+ size = omap_mcbsp_get_fifo_size(mcbsp_data->bus_id);
+
+ frames.min = size / channels->min;
+ frames.integer = 1;
+ return snd_interval_refine(buffer_size, &frames);
}
static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
@@ -182,33 +214,35 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
if (!cpu_dai->active)
err = omap_mcbsp_request(bus_id);
+ /*
+ * OMAP3 McBSP FIFO is word structured.
+ * McBSP2 has 1024 + 256 = 1280 word long buffer,
+ * McBSP1,3,4,5 has 128 word long buffer
+ * This means that the size of the FIFO depends on the sample format.
+ * For example on McBSP3:
+ * 16bit samples: size is 128 * 2 = 256 bytes
+ * 32bit samples: size is 128 * 4 = 512 bytes
+ * It is simpler to place constraint for buffer and period based on
+ * channels.
+ * McBSP3 as example again (16 or 32 bit samples):
+ * 1 channel (mono): size is 128 frames (128 words)
+ * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
+ * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
+ */
if (cpu_is_omap343x()) {
- int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
- int max_period;
-
/*
- * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
- * Set constraint for minimum buffer size to the same than FIFO
- * size in order to avoid underruns in playback startup because
- * HW is keeping the DMA request active until FIFO is filled.
- */
- if (bus_id == 1)
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
- 4096, UINT_MAX);
-
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
- else
- max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
-
- max_period++;
- max_period <<= 1;
-
- if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
- 32, max_period);
+ * Rule for the buffer size. We should not allow
+ * smaller buffer than the FIFO size to avoid underruns
+ */
+ snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ omap_mcbsp_hwrule_min_buffersize,
+ mcbsp_data,
+ SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1);
+
+ /* Make sure, that the period size is always even */
+ snd_pcm_hw_constraint_step(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
}
return err;
@@ -289,11 +323,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
- int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
+ struct omap_pcm_dma_data *dma_data;
+ int dma, bus_id = mcbsp_data->bus_id;
int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
+ int pkt_size = 0;
unsigned long port;
unsigned int format, div, framesize, master;
+ dma_data = &omap_mcbsp_dai_dma_params[cpu_dai->id][substream->stream];
if (cpu_class_is_omap1()) {
dma = omap1_dma_reqs[bus_id][substream->stream];
port = omap1_mcbsp_port[bus_id][substream->stream];
@@ -306,35 +343,74 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
} else if (cpu_is_omap343x()) {
dma = omap24xx_dma_reqs[bus_id][substream->stream];
port = omap34xx_mcbsp_port[bus_id][substream->stream];
- omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
- omap_mcbsp_set_threshold;
- /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
- if (omap_mcbsp_get_dma_op_mode(bus_id) ==
- MCBSP_DMA_MODE_THRESHOLD)
- sync_mode = OMAP_DMA_SYNC_FRAME;
} else {
return -ENODEV;
}
- omap_mcbsp_dai_dma_params[id][substream->stream].name =
- substream->stream ? "Audio Capture" : "Audio Playback";
- omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
- omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
- omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
- omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
- OMAP_DMA_DATA_TYPE_S16;
+ dma_data->data_type = OMAP_DMA_DATA_TYPE_S16;
+ wlen = 16;
break;
case SNDRV_PCM_FORMAT_S32_LE:
- omap_mcbsp_dai_dma_params[id][substream->stream].data_type =
- OMAP_DMA_DATA_TYPE_S32;
+ dma_data->data_type = OMAP_DMA_DATA_TYPE_S32;
+ wlen = 32;
break;
default:
return -EINVAL;
}
+ if (cpu_is_omap343x()) {
+ dma_data->set_threshold = omap_mcbsp_set_threshold;
+ /* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+ if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+ MCBSP_DMA_MODE_THRESHOLD) {
+ int period_words, max_thrsh;
+
+ period_words = params_period_bytes(params) / (wlen / 8);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ max_thrsh = omap_mcbsp_get_max_tx_threshold(
+ mcbsp_data->bus_id);
+ else
+ max_thrsh = omap_mcbsp_get_max_rx_threshold(
+ mcbsp_data->bus_id);
+ /*
+ * If the period contains less or equal number of words,
+ * we are using the original threshold mode setup:
+ * McBSP threshold = sDMA frame size = period_size
+ * Otherwise we switch to sDMA packet mode:
+ * McBSP threshold = sDMA packet size
+ * sDMA frame size = period size
+ */
+ if (period_words > max_thrsh) {
+ int divider = 0;
+
+ /*
+ * Look for the biggest threshold value, which
+ * divides the period size evenly.
+ */
+ divider = period_words / max_thrsh;
+ if (period_words % max_thrsh)
+ divider++;
+ while (period_words % divider &&
+ divider < period_words)
+ divider++;
+ if (divider == period_words)
+ return -EINVAL;
+
+ pkt_size = period_words / divider;
+ sync_mode = OMAP_DMA_SYNC_PACKET;
+ } else {
+ sync_mode = OMAP_DMA_SYNC_FRAME;
+ }
+ }
+ }
- snd_soc_dai_set_dma_data(cpu_dai, substream,
- &omap_mcbsp_dai_dma_params[id][substream->stream]);
+ dma_data->name = substream->stream ? "Audio Capture" : "Audio Playback";
+ dma_data->dma_req = dma;
+ dma_data->port_addr = port;
+ dma_data->sync_mode = sync_mode;
+ dma_data->packet_size = pkt_size;
+
+ snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
if (mcbsp_data->configured) {
/* McBSP already configured by another stream */
@@ -360,7 +436,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
/* Set word lengths */
- wlen = 16;
regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16);
regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16);
regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16);
@@ -368,7 +443,6 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
break;
case SNDRV_PCM_FORMAT_S32_LE:
/* Set word lengths */
- wlen = 32;
regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32);
regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32);
regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32);
@@ -409,6 +483,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
}
omap_mcbsp_config(bus_id, &mcbsp_data->regs);
+ mcbsp_data->wlen = wlen;
mcbsp_data->configured = 1;
return 0;
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 87ce842fa2e8..9eecac135bbb 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -43,12 +43,14 @@
static struct regulator *omap3pandora_dac_reg;
-static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params, unsigned int fmt)
+static int omap3pandora_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
int ret;
/* Set codec DAI configuration */
@@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
return 0;
}
-static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- return omap3pandora_cmn_hw_params(substream, params,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_IB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
-}
-
-static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- return omap3pandora_cmn_hw_params(substream, params,
- SND_SOC_DAIFMT_I2S |
- SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS);
-}
-
static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
@@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec)
return snd_soc_dapm_sync(codec);
}
-static struct snd_soc_ops omap3pandora_out_ops = {
- .hw_params = omap3pandora_out_hw_params,
-};
-
-static struct snd_soc_ops omap3pandora_in_ops = {
- .hw_params = omap3pandora_in_hw_params,
+static struct snd_soc_ops omap3pandora_ops = {
+ .hw_params = omap3pandora_hw_params,
};
/* Digital audio interface glue - connects codec <--> CPU */
@@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = {
.stream_name = "HiFi Out",
.cpu_dai = &omap_mcbsp_dai[0],
.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
- .ops = &omap3pandora_out_ops,
+ .ops = &omap3pandora_ops,
.init = omap3pandora_out_init,
}, {
.name = "TWL4030",
.stream_name = "Line/Mic In",
.cpu_dai = &omap_mcbsp_dai[1],
.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
- .ops = &omap3pandora_in_ops,
+ .ops = &omap3pandora_ops,
.init = omap3pandora_in_init,
}
};
diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c
index 47d831ef2dbb..88052d29617f 100644
--- a/sound/soc/omap/rx51.c
+++ b/sound/soc/omap/rx51.c
@@ -27,6 +27,7 @@
#include <linux/gpio.h>
#include <linux/platform_device.h>
#include <sound/core.h>
+#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
@@ -37,14 +38,22 @@
#include "omap-pcm.h"
#include "../codecs/tlv320aic3x.h"
+#define RX51_TVOUT_SEL_GPIO 40
+#define RX51_JACK_DETECT_GPIO 177
/*
* REVISIT: TWL4030 GPIO base in RX-51. Now statically defined to 192. This
* gpio is reserved in arch/arm/mach-omap2/board-rx51-peripherals.c
*/
#define RX51_SPEAKER_AMP_TWL_GPIO (192 + 7)
+enum {
+ RX51_JACK_DISABLED,
+ RX51_JACK_TVOUT, /* tv-out */
+};
+
static int rx51_spk_func;
static int rx51_dmic_func;
+static int rx51_jack_func;
static void rx51_ext_control(struct snd_soc_codec *codec)
{
@@ -57,6 +66,9 @@ static void rx51_ext_control(struct snd_soc_codec *codec)
else
snd_soc_dapm_disable_pin(codec, "DMic");
+ gpio_set_value(RX51_TVOUT_SEL_GPIO,
+ rx51_jack_func == RX51_JACK_TVOUT);
+
snd_soc_dapm_sync(codec);
}
@@ -162,6 +174,40 @@ static int rx51_set_input(struct snd_kcontrol *kcontrol,
return 1;
}
+static int rx51_get_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = rx51_jack_func;
+
+ return 0;
+}
+
+static int rx51_set_jack(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (rx51_jack_func == ucontrol->value.integer.value[0])
+ return 0;
+
+ rx51_jack_func = ucontrol->value.integer.value[0];
+ rx51_ext_control(codec);
+
+ return 1;
+}
+
+static struct snd_soc_jack rx51_av_jack;
+
+static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
+ {
+ .gpio = RX51_JACK_DETECT_GPIO,
+ .name = "avdet-gpio",
+ .report = SND_JACK_VIDEOOUT,
+ .invert = 1,
+ .debounce_time = 200,
+ },
+};
+
static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
SND_SOC_DAPM_MIC("DMic", NULL),
@@ -177,10 +223,12 @@ static const struct snd_soc_dapm_route audio_map[] = {
static const char *spk_function[] = {"Off", "On"};
static const char *input_function[] = {"ADC", "Digital Mic"};
+static const char *jack_function[] = {"Off", "TV-OUT"};
static const struct soc_enum rx51_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
};
static const struct snd_kcontrol_new aic34_rx51_controls[] = {
@@ -188,10 +236,13 @@ static const struct snd_kcontrol_new aic34_rx51_controls[] = {
rx51_get_spk, rx51_set_spk),
SOC_ENUM_EXT("Input Select", rx51_enum[1],
rx51_get_input, rx51_set_input),
+ SOC_ENUM_EXT("Jack Function", rx51_enum[2],
+ rx51_get_jack, rx51_set_jack),
};
static int rx51_aic34_init(struct snd_soc_codec *codec)
{
+ struct snd_soc_card *card = codec->socdev->card;
int err;
/* Set up NC codec pins */
@@ -214,7 +265,16 @@ static int rx51_aic34_init(struct snd_soc_codec *codec)
snd_soc_dapm_sync(codec);
- return 0;
+ /* AV jack detection */
+ err = snd_soc_jack_new(card, "AV Jack",
+ SND_JACK_VIDEOOUT, &rx51_av_jack);
+ if (err)
+ return err;
+ err = snd_soc_jack_add_gpios(&rx51_av_jack,
+ ARRAY_SIZE(rx51_av_jack_gpios),
+ rx51_av_jack_gpios);
+
+ return err;
}
/* Digital audio interface glue - connects codec <--> CPU */
@@ -259,6 +319,11 @@ static int __init rx51_soc_init(void)
if (!machine_is_nokia_rx51())
return -ENODEV;
+ err = gpio_request(RX51_TVOUT_SEL_GPIO, "tvout_sel");
+ if (err)
+ goto err_gpio_tvout_sel;
+ gpio_direction_output(RX51_TVOUT_SEL_GPIO, 0);
+
rx51_snd_device = platform_device_alloc("soc-audio", -1);
if (!rx51_snd_device) {
err = -ENOMEM;
@@ -277,13 +342,19 @@ static int __init rx51_soc_init(void)
err2:
platform_device_put(rx51_snd_device);
err1:
+ gpio_free(RX51_TVOUT_SEL_GPIO);
+err_gpio_tvout_sel:
return err;
}
static void __exit rx51_soc_exit(void)
{
+ snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
+ rx51_av_jack_gpios);
+
platform_device_unregister(rx51_snd_device);
+ gpio_free(RX51_TVOUT_SEL_GPIO);
}
module_init(rx51_soc_init);
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-22 08:38:13 +0100